similar to: Asterisk 1.4.26.2 died after 80 days uptime

Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas

I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

from extensions.conf: exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) I basically try to lookup the CLIP and attach a name for each inbound call. This works fine, except when I have just restarted asterisk - at which time I've more than once seen the message from the subject. As far as I can tell, with my Set(CALLERID), I should always have an argument in the DB

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I've got a Grandstream 487 in a home-office. The phone-side is working fine, but the user is complaining that his internet connection keeps disappearing. The Grandstream is set up as NAT router, and there's just one PC hanging off the LAN. Has anyone experienced anything similar? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get "module reload app_queue" executed automatically at startup?

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. /Per Jessen, Z?rich -- http://www.spamchek.com/ - managed email security.

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [4439000@Business:3] Queue("mISDN/3-u0", "enidan|t|||20") in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

Per Jessen wrote: > Per Jessen wrote: > >> Michael Kliewe wrote: >> >>> Hi Per, >>> >>> On 25.08.2015 11:40, Per Jessen wrote: >>>> sieve and managesieve are both working fine, have been for years. >>>> Now I want to add a sieve_after script: >>>> >>>> /etc/dovecot/conf.d/90-sieve.conf :

Environment: dovecot 2.0.9, postfix 2.6, Pigeonhole 0.2.2 I'm in the process of setting up vacation auto-reply for the first time, (it's that time of the year) and it looks like .dovecot.lda-dupes is being updated, but not read (or just ignored). my vacation script: require ["vacation","variables"]; # get hold of the subject if header :matches "Subject"

Michael Kliewe wrote: > Hi Per, > > On 25.08.2015 11:40, Per Jessen wrote: >> sieve and managesieve are both working fine, have been for years. Now >> I want to add a sieve_after script: >> >> /etc/dovecot/conf.d/90-sieve.conf : >> >> plugin { >> sieve = ~/.dovecot.sieve >> sieve_dir = ~/sieve >> sieve_before = >>

This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ?, ?, ?, ?, ?, ? etc. ? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Hi, I have just started to get dovecot list messages which I had not been receiving until today. How can I opt out (again)? regards, Michal Szymanski On Fri, Nov 30, 2018 at 10:42:22AM +0100, Per Jessen wrote: > Aki Tuomi wrote: > >On 30.11.2018 10.03, Per Jessen wrote: > > > >>Hi AKi > >> > >>I guess my address was re-subscribed then? I was subscribed

Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: